Early telephone systems had a separate circuit from the central office to each telephone and a switchboard with plugs and wires in the central office for a person to make connections. Automated switches replaced the plugs and wires operated by a human. Then some of the automated switching functions were moved upstream into private equipment in company offices to save the cost of requiring a separate line from each telephone to the central office. Because the central office was called an “exchange”, the remote switch was called a “private branch exchange”, or PBX. At first, the PBX merely performed switching functions. Then, as additional functions were invented for business telephones, such as hold, transfer, conference, indicator lights, and displays, what was merely a switch became what we refer to here as a “call controller”. A KTS (Key Telephone System) is a simpler and less expensive form of such a call controller. Within this document, the term “call controller” is used to describe both such PBX and KTS devices as well as a newer form of call controller that uses packet switching on a packet switched network, preferably a network running Internet Protocol (IP), discussed further below. Although the new IP call controller is “private”, it is not in any sense a “branch exchange”, so the term IP-PBX, while it is commonly used, is misdescriptive.
The handsets that are used with a PBX or a KTS support many additional features that a standard analog telephone for use with the public network cannot support. Although there are differences in functionality between a PBX and a KTS, the handsets for use with either are essentially the same. When this document refers to a PBX telephone handset, it means a handset with additional features for use with a PBX or with a KTS that supports those features. Likewise, when this document refers to PBX equipment, unless the context requires otherwise, it means PBX or KTS equipment.
Standard PBX (or KTS) equipment for the last 30 years has used digital communications between the PBX and the telephone handset to exchange the various call control signals between the PBX call controller and the digital handset. Within the PBX, in order for a circuit to handle control signaling and many telephone call voice signals at once, time division multiplexing (TDM) was developed. In this method, each hundredth of a second is divided into many much smaller time slots and the time slots are allocated sequentially among many circuits. Each circuit receives a high enough percentage of each hundredth of a second of time to produce a voice quality connection for reception by a human.
Although digital signaling between each handset and the PBX with TDM multiplexing in the PBX has become universally adopted by all telephone equipment manufacturers, there are no standards. Consequently, the handset of one manufacturer will not operate with the PBX of another manufacturer. However, as publicly regulated monopolies, telephone companies began to offer similar call control functions from their central offices with digital call control signaling from call controllers running public signaling protocols called “Centrex” systems. Centrex systems and PBX systems all used “circuit switched” networks where a single circuit, although it may be time division multiplexed, is established between each pair of telephones in communication with each other and those telephones use 100 percent of that circuit, even when neither party is talking.
Economies of scale can be accomplished if the voice communications are merged with data communications, and further economies are achieved if the communications are sent via packet switched networks rather than circuit switched networks. A packet switched network can merge packets from many different origins destined for many different destinations into a single “channel” and then separately switch them to different directions at a subsequent point in their journey, as opposed to a circuit switched PBX (or KTS) which switches calls through the creation of physical electronic circuits and uses time division multiplexing (TDM) on a local area bus.
Attempts to merge voice communications and data communications over packet switched networks did not achieve market acceptance until implementation of the global computer network based on Internet Protocol (IP). IP telephony is revolutionizing the telecom industry with promises of the following benefits (among others):                1. Eliminating distance sensitivity in pricing and telecom features. A call from London to Seattle can cost the same and provide the same features as a call between two offices on the same floor.        2. Easing the development and deployment of intelligent features such as computer-telephone integration, “reach me” and “personal assistant” services, and unified messaging.        3. Reducing the cost of telephone systems by leveraging the economies of scale that come from putting voice and data traffic on a single data network rather than two disparate and separately maintained networks (one for voice, one for data).        
These benefits of IP telephony are typically delivered through a proprietary IP call controller (often misdescriptively called an “IP-PBX”—Private Branch Exchange—by analogy to a traditional PBX) using packet switching on a data network. Some of the same benefits can also be provided by a public IP call controller, an IP Centrex system.
To obtain some of the benefits of using the global IP network for telephone communications, as shown in FIG. 3, PBX manufacturers have been adding an internet protocol interface on the trunk side of the PBX 51 so that, in addition to sending out calls over the public switched telephone network (PSTN) 15, they can also send calls via Voice over IP (VoIP) on the Internet 23. However, further benefits can be achieved by communicating between the handsets and the call controller via IP on a data network using IP handsets and IP call controllers (IP-PBXs).
Most existing circuit switched PBXs (and KTSs) support desktop telephone handsets which use a variety of proprietary digital signaling methods to deliver enhanced features such as an LCD call status display, multiple line appearances, various indicator lamps, and intelligent “feature buttons”. Contemporary IP PBX systems do not support these handsets but instead support proprietary IP digital telephone handsets. The IP digital telephone handsets connect directly to an IP network and therefore require fairly intelligent IP circuitry in the handset, which makes them rather expensive compared to the digital telephones employed by a typical circuit PBX. Most IP PBX's also support the attachment of standard analog telephones via various gateway devices. This reduces the cost of the handsets, but analog telephones do not support the advanced features which a typical business user expects—such as LCD display and indicator lights. So the trade-off for users of IP telephony is: a) pay more for IP phones with lots of features, or b) settle for less expensive analog phones with gateways but fewer features.
A hybrid has been developed that allows companies to use their existing handsets and traditional PBX equipment but carry the communications between the two across an IP network so that the handsets and PBX can be located remotely from each other while using the IP network to achieve a highly effective low cost connection. As shown in FIG. 4, this system requires a gateway 62 that places the PBX signals into packets and provides IP headers for those packets so they can be transmitted on an IP network. Similarly, the handsets 10 are coupled to a remote access interface 61 that receives packets from the PBX, extracts the voice data and feature signaling data from each packet, and forwards the resulting PBX type data to the appropriate handset. Likewise, it receives signaling data from each handset and encapsulates the data into IP packets, adds a header to each packet, and sends them on the network to the gateway where the data is decapsulated.
When voice communications (or other real time communications) are sent over an IP network, the packets must be given precedence over packets that are not sensitive to real time delivery, such as computer data packets, to avoid problems of perceptible time delays and “jitter” which is a disruption in voice quality resulting from otherwise imperceptible time delays. Therefore, voice communications are routed over IP network connections where such precedence can be managed over each link in the network.